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- /*
- * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
- #ifndef MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
- #define MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
- #include <vector>
- #include "absl/types/optional.h"
- #include "system_wrappers/include/clock.h"
- namespace webrtc {
- namespace test {
- class PerformanceTimer {
- public:
- explicit PerformanceTimer(int num_frames_to_process);
- ~PerformanceTimer();
- void StartTimer();
- void StopTimer();
- double GetDurationAverage() const;
- double GetDurationStandardDeviation() const;
- // These methods are the same as those above, but they ignore the first
- // |number_of_warmup_samples| measurements.
- double GetDurationAverage(size_t number_of_warmup_samples) const;
- double GetDurationStandardDeviation(size_t number_of_warmup_samples) const;
- private:
- webrtc::Clock* clock_;
- absl::optional<int64_t> start_timestamp_us_;
- std::vector<int64_t> timestamps_us_;
- };
- } // namespace test
- } // namespace webrtc
- #endif // MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
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